US7643988B2 - Method for analyzing fundamental frequency information and voice conversion method and system implementing said analysis method - Google Patents

Method for analyzing fundamental frequency information and voice conversion method and system implementing said analysis method Download PDF

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US7643988B2
US7643988B2 US10/551,224 US55122404A US7643988B2 US 7643988 B2 US7643988 B2 US 7643988B2 US 55122404 A US55122404 A US 55122404A US 7643988 B2 US7643988 B2 US 7643988B2
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fundamental frequency
spectral envelope
voice
speaker
spectral
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US20060178874A1 (en
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Taoufik En-Najjary
Olivier Rosec
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Orange SA
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France Telecom SA
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/90—Pitch determination of speech signals
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/003—Changing voice quality, e.g. pitch or formants
    • G10L21/007—Changing voice quality, e.g. pitch or formants characterised by the process used
    • G10L21/013—Adapting to target pitch
    • G10L2021/0135—Voice conversion or morphing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G10L25/24—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being the cepstrum

Definitions

  • the present invention relates to a method for analyzing fundamental frequency information contained in voice samples, and a voice conversion method and system implementing said analysis method.
  • production of speech may entail vibration of the vocal chords, which manifests itself through the presence in the speech signal of a periodic structure having a fundamental period, the inverse of which is referred to as the fundamental frequency or pitch.
  • aural rendering is of vital importance, and effective control of the parameters linked to prosody, including the fundamental frequency, is required in order to obtain acceptable quality.
  • the object of the present invention is to overcome this problem by defining a method for analyzing fundamental frequency information of voice samples, making it possible to define a fundamental frequency representation whose parameters can be defined.
  • the subject of the present invention is a method for analyzing fundamental frequency information contained in voice samples, characterized in that it comprises at least:
  • the invention also relates to a method for the conversion of a voice signal pronounced by a source speaker into a converted voice signal whose characteristics resemble those of a target speaker, comprising at least:
  • the invention also relates to a system for converting a voice signal pronounced by a source speaker into a converted voice signal whose characteristics resemble those of a target speaker, said system comprising at least:
  • FIG. 1 is a flowchart of an analysis method according to the invention
  • FIG. 2 is a flowchart of a voice conversion method implementing the analysis method according to the invention.
  • FIG. 3 is a functional block diagram of a voice conversion system, enabling the implementation of the method according to the invention described in FIG. 2 .
  • the method according to the invention shown in FIG. 1 is implemented on the basis of a database of voice samples containing sequences of natural speech.
  • the method starts with a step 2 for analyzing samples by grouping them together in frames, in order to obtain, for each sample frame, spectrum-related information and, in particular, information relating to the spectral envelope, and information relating to the fundamental frequency.
  • this analysis step 2 is based on the use of a model of a sound signal in the form of a sum of a harmonic signal and a noise signal according to a model normally referred to as “HNM” (Harmonic plus Noise Model).
  • HNM Harmonic plus Noise Model
  • the embodiment described is based on a representation of the spectral envelope by the discrete cepstrum.
  • a cepstral representation in fact enables separation, in the speech signal, of the component relating to the vocal tract from the resulting source component, corresponding to the vibrations of the vocal chords and characterized by the fundamental frequency.
  • analysis step 2 comprises a sub-step 4 for modeling each voice signal frame into a harmonic part representing the periodic component of the signal, consisting of a sum of L harmonic sinusoids with amplitude A
  • s ⁇ ( n ) h ⁇ ( n ) + b ⁇ ( n )
  • h(n) therefore represents the harmonic approximation of the signal s(n).
  • Step 2 then comprises a sub-step 5 for estimating, for each frame, frequency parameters, of the fundamental frequency in particular, for example by means of an autocorrelation method.
  • this HNM analysis supplies the maximum voicing frequency.
  • this frequency may be arbitrarily defined, or may be estimated by other known means.
  • This sub-step 5 is followed by a sub-step 6 for synchronized analysis of the fundamental frequency of each frame, enabling estimation of the parameters of the harmonic part and the parameters of the signal noise.
  • this synchronized analysis corresponds to the determination of the harmonic parameters through minimization of a weighted least squares criterion between the full signal and its harmonic breakdown, corresponding, in the embodiment described, to the estimated noise signal.
  • the criterion denoted as E is equal to:
  • w(n) is the analysis window and T i is the fundamental period of the current frame.
  • the analysis window is centered around the fundamental period marker and its duration is twice this period.
  • the analysis step 2 lastly comprises a sub-step 7 for estimating the parameters of the components of the spectral envelope of the signal, using, for example, a regularized discrete cepstrum method and a Bark-scale transformation in order to reproduce the properties of the human ear as faithfully as possible.
  • the analysis step 2 supplies, for each frame of order n of speech signal samples, a scalar denoted as x n , comprising fundamental frequency information, and a vector denoted as y n , comprising spectral information in the form of a sequence of cepstral coefficients.
  • the analysis step 2 is followed by a step 10 for normalizing the value of the fundamental frequency of each frame in relation to the mean fundamental frequency in order to replace, in each voice sample frame, the value of the fundamental frequency with a fundamental frequency value normalized according to the following formula:
  • F log log ⁇ ⁇ ( F o F o moy )
  • F o moy corresponds to the mean of the values of the fundamental frequencies over the entire analyzed database.
  • This normalization enables modification of the scale of the variations of the fundamental frequency scalars in order to make it consistent with the scale of the cepstral coefficient variations.
  • the normalization step 10 is followed by a step 20 for determining a model representing the common cepstrum and fundamental frequency characteristics of all the analyzed samples.
  • the embodiment described involves a probabilistic model of the fundamental frequency and of the discrete cepstrum according to a Gaussian densities mixture model, generally referred to as “GMM”, the parameters of which are estimated on the basis of the joint density of the normalized fundamental frequency and the discrete cepstrum.
  • GMM Gaussian densities mixture model
  • the probability density of a random variable denoted in a general manner as p(z), according to a Gaussian densities mixture model GMM, is denoted mathematically in the following manner:
  • N(z: ⁇ i ; ⁇ i ) is the probability density of the normal law of mean ⁇ i and the covariance matrix ⁇ i and the coefficients ⁇ i are the coefficients of the mixture.
  • the coefficient ⁇ i corresponds to the a priori probability that the random variable z is generated by the i th Gaussian of the mixture.
  • the step 20 for determining the model comprises a sub-step 22 for modeling the joint density of the cepstrum denoted as y and the normalized fundamental frequency denoted as x, in such a way that:
  • the step 20 then comprises a sub-step 24 for estimating GMM parameters ( ⁇ , ⁇ , ⁇ ) of the density p(z).
  • This estimation may be implemented, for example, with the aid of a conventional algorithm of the type known as “EM” (Expectation Maximization), corresponding to an iterative method by means of which an estimator of the maximum resemblance between the speech sample data and the Gaussian mixture model is obtained.
  • the determination of the initial parameters of the GMM model is obtained with the aid of a conventional vector quantification technique.
  • the model determination step 20 thus supplies the parameters of a mixture of Gaussian densities representing common spectral characteristics, represented by the cepstrum coefficients, and fundamental frequencies of the analyzed voice samples.
  • the method then comprises a step 30 for determining, on the basis of the model and voice samples, a fundamental frequency prediction function exclusively according to spectral information supplied by the signal cepstrum.
  • This prediction function is determined on the basis of an estimator of the implementation of the fundamental frequency, given the cepstrum of the voice samples, formed in the embodiment described by the conditional expectation.
  • the step 30 comprises a sub-step 32 for determining the conditional expectation of the fundamental frequency, knowing the spectrum-related information supplied by the cepstrum.
  • the conditional expectation is denoted as F(y) and is determined on the basis of the following formulae:
  • P i (y) corresponds to the a posteriori probability that the cepstrum vector y is generated by the i th component of the Gaussian mixture of the model, defined in step 20 by the covariance matrix ⁇ i and the normal law ⁇ i .
  • the determination of the conditional expectation thus enables the fundamental frequency prediction function to be obtained from the cepstrum information.
  • the estimator implemented in step 30 may be an a posteriori maximum criterion, referred to as “MAP”, and corresponding to the implementation of the expectation calculation exclusively for the model best representing the source vector.
  • MAP a posteriori maximum criterion
  • the analysis method according to the invention enables, on the basis of the model and the voice samples, a fundamental frequency prediction function to be obtained exclusively according to spectral information supplied, in the embodiment described, by the cepstrum.
  • a prediction function of this type then enables the fundamental frequency value for a speech signal to be determined exclusively on the basis of spectral information of this signal, thereby enabling a relevant prediction of the fundamental frequency, in particular for sounds which are not in the analyzed voice samples.
  • Voice conversion consists in modifying the voice signal of a reference speaker known as the “source speaker” in such a way that the signal produced appears to have been pronounced by a different speaker referred to as the “target speaker”.
  • This method is implemented using a database of voice samples pronounced by the source speaker and the target speaker.
  • a method of this type comprises a step 50 for determining a transformation function for the spectral characteristics of the voice samples of the source speaker to make them resemble the spectral characteristics of the voice samples of the target speaker.
  • this step 50 is based on an HNM analysis which enables the relationships between the characteristics of the spectral envelope of the voice signals of the source and target speakers to be determined.
  • Source and target voice recordings corresponding to the acoustic realization of the same phonetic sequence are required for this purpose.
  • the step 50 comprises a sub-step 52 for modeling voice samples according to an HNM sum model of harmonic and noise signals.
  • the sub-step 52 is followed by a sub-step 54 enabling alignment of the source and target signals with the aid, for example, of a conventional alignment algorithm known as “DTW” (Dynamic Time Warping).
  • DTW Dynamic Time Warping
  • Step 50 then comprises a sub-step 56 for determining a model such as a GMM model representing the common characteristics of the voice sample spectra of the source and target speakers.
  • a model such as a GMM model representing the common characteristics of the voice sample spectra of the source and target speakers.
  • a GMM model which comprises 64 components and a single vector containing the cepstral parameters of the source and target, in such a way that a spectral transformation function can be defined which corresponds to an estimator of the realization of the target spectral parameters denoted as t, knowing the source spectral parameters denoted as s.
  • this transformation function denoted as F(s) is denoted in the form of a conditional expectation obtained by the following formula:
  • the estimator may be formed from an a posteriori maximum criterion.
  • the function thus defined therefore enables modification of the spectral envelope of a speech signal originating from the source speaker in order to make it resemble the spectral envelope of the target speaker.
  • the parameters of the GMM model representing the common spectral characteristics of the source and target are initialized, for example, with the aid of a vector quantification algorithm.
  • the analysis method according to the invention is implemented in a step 60 in which only the voice samples of the target speaker are analyzed.
  • the analysis step 60 enables a fundamental frequency prediction function to be obtained for the target speaker, exclusively on the basis of spectral information.
  • the conversion method then comprises a step 65 in which a voice signal to be converted, pronounced by the source speaker, is analyzed, said signal to be converted being different from the voice signals used in steps 50 and 60 .
  • This analysis step 65 is implemented, for example, with the aid of a breakdown according to the HNM model, enabling the provision of spectral information in the form of cepstral coefficients, fundamental frequency information and maximum frequency and phase voicing information.
  • This step 65 is followed by a step 70 in which the spectral characteristics of the voice signal to be converted are transformed by applying the transformation function determined in step 50 to the cepstral coefficients defined in step 65 .
  • This step 70 modifies the spectral envelope of the voice signal to be converted.
  • each frame of samples of the source speaker signal to be converted is thus associated with transformed spectral information whose characteristics are similar to the spectral characteristics of the samples of the target speaker.
  • the conversion method then comprises a fundamental frequency prediction step 80 for the voice samples of the source speaker, by applying the prediction function determined using the method according to the invention in step 60 , exclusively to the transformed spectral information associated with the source speaker voice signal to be converted.
  • the prediction function defined in step 60 enables a relevant prediction of the fundamental frequency to be obtained.
  • the conversion method then comprises an output signal synthesis step 90 , implemented, in the example described, by an HNM synthesis which directly supplies the voice signal converted on the basis of the transformed spectral envelope information supplied in step 70 , the predicted fundamental frequency information produced in step 80 and the maximum frequency and phase voicing information supplied by step 65 .
  • the conversion method implementing the analysis method according to the invention thus enables a voice conversion to be obtained which implements spectral modifications and a fundamental frequency prediction in such a way as to obtain a high-quality aural rendering.
  • the effectiveness of a method of this type can be evaluated on the basis of identical voice samples pronounced by the source speaker and the target speaker.
  • the voice signal pronounced by the source speaker is converted with the aid of the method as described, and the resemblance between the converted signal and the signal pronounced by the target speaker is evaluated.
  • this resemblance is calculated in the form of a ratio between the acoustic distance separating the converted signal from the target signal and the acoustic distance separating the target signal from the source signal.
  • the ratio obtained for a signal converted with the aid of the method according to the invention is in the order of 0.3 to 0.5.
  • FIG. 3 shows a functional block diagram of a voice conversion system implementing the method described with reference to FIG. 2 .
  • This system uses at its input a database 100 of voice samples pronounced by the source speaker and a database 102 containing at least the same voice samples pronounced by the target speaker.
  • a module 104 which determines a function for transforming spectral characteristics of the source speaker into spectral characteristics of the target speaker.
  • This module 104 is adapted for the implementation of step 50 of the method as described with reference to FIG. 2 , and therefore enables the determination of a spectral envelope transformation function.
  • the system comprises a module 106 for determining a fundamental frequency prediction function exclusively according to spectrum-related information.
  • the module 106 receives at its input voice samples of the target speaker only, contained in the database 102 .
  • the module 106 is adapted for the implementation of step 60 of the method described with reference to FIG. 2 , corresponding to the analysis method according to the invention as described with reference to FIG. 1 .
  • the transformation function supplied by the module 104 and the prediction function supplied by the module 106 are advantageously stored with a view to subsequent use.
  • the voice conversion system receives at its input a voice signal 110 corresponding to a speech signal pronounced by the source speaker and intended to be converted.
  • the signal 110 is introduced into a signal analysis module 112 , implementing, for example, an HNM breakdown and enabling dissociation of the spectral information of the signal 110 in the form of cepstral coefficients and fundamental frequency information.
  • the module 112 also supplies maximum frequency and phase voicing information obtained by applying the HNM model.
  • the module 112 therefore implements the step 65 of the method previously described.
  • This analysis may possibly be carried out in advance, and the information is stored for subsequent use.
  • the cepstral coefficients supplied by the module 112 are then introduced into a transformation module 114 adapted to apply the transformation function determined by the module 104 .
  • the transformation module 114 implements step 70 of the method described with reference to FIG. 2 and supplies the transformed cepstral coefficients whose characteristics are similar to the spectral characteristics of the target speaker.
  • the module 114 thus implements a modification of the spectral envelope of the voice signal 110 .
  • the transformed cepstral coefficients supplied by the module 114 are then introduced into a fundamental frequency prediction module 116 adapted to implement the prediction function determined by the module 106 .
  • the module 116 implements step 80 of the method described with reference to FIG. 2 and supplies at its output fundamental frequency information predicted exclusively on the basis of the transformed spectral information.
  • the system then comprises a synthesis module 118 receiving at its input the transformed cepstral coefficients originating from the module 114 and corresponding to the spectral envelope, the predicted fundamental frequency information originating from the module 116 , and the maximum frequency and phase voicing information supplied by the module 112 .
  • the module 118 thus implements step 90 of the method described with reference to FIG. 2 and supplies a signal 120 corresponding to the voice signal 110 of the source speaker, except that its spectral and fundamental frequency characteristics have been modified in order to be similar to those of the target speaker.
  • the system described may be implemented in various ways, in particular with the aid of a suitable computer program connected to sound acquisition hardware means.
  • HNM and GMM models may be replaced by other techniques and models known to the person skilled in the art, such as, for example, LSF (Line Spectral Frequencies) and LPC (Linear Predictive Coding) techniques, or format-related parameters.
  • LSF Line Spectral Frequencies
  • LPC Linear Predictive Coding

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FR03/03790 2003-03-27
FR0303790A FR2853125A1 (fr) 2003-03-27 2003-03-27 Procede d'analyse d'informations de frequence fondamentale et procede et systeme de conversion de voix mettant en oeuvre un tel procede d'analyse.
PCT/FR2004/000483 WO2004088633A1 (fr) 2003-03-27 2004-03-02 Procede d'analyse d'informations de frequence fondamentale et procede et systeme de conversion de voix mettant en oeuvre un tel procede d'analyse

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WO2018138543A1 (en) * 2017-01-24 2018-08-02 Hua Kanru Probabilistic method for fundamental frequency estimation

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CN102664003B (zh) * 2012-04-24 2013-12-04 南京邮电大学 基于谐波加噪声模型的残差激励信号合成及语音转换方法
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US8010362B2 (en) * 2007-02-20 2011-08-30 Kabushiki Kaisha Toshiba Voice conversion using interpolated speech unit start and end-time conversion rule matrices and spectral compensation on its spectral parameter vector
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JP2006521576A (ja) 2006-09-21
ATE395684T1 (de) 2008-05-15
JP4382808B2 (ja) 2009-12-16
WO2004088633A1 (fr) 2004-10-14
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FR2853125A1 (fr) 2004-10-01
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